[Asterisk-Users] Calls to Turkey, any good providers?
Hello All once again... Has anyone got any experience with calling to Turkey? Voipjet seems to have good quality and rates, but I was wondering if there are any termination providers over there, or providers that can supply a DID, even in a home-user scenario. Thanks, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice europe plus calling plan quality
Hello All, I am trying broadvoice's europe plus calling plan for unlimited to Poland. My first attempts though, were not that good. I could hear the other side, but they could not clearly hear me. Is this because broadvoice's connection just is not up to par? Has anyone else been using this plan to europe and had success? Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 versus PRI
Hello All, I was wondering. What are the primary advantages to using a PRI over a T1? As I understand it, the PRI terminates very fast, meaning you can do immediate answer and dial... This is very handy on the BRI line I have on the asterisk. Can T1 signalling also do immediate answer, or does it just behave like a channelized pots line and ring as usual? I am trying to determine if I should use a pri between a pbx and asterisk, but PRI costs about double to implement. I do however enjoy the fast connect time... Thanks, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] URGENT: Problems with PHP AGI...
A) AGI prefers the CLI version. B) Use VERBOSE, write to stderr or dump any debug messages in your own log file C) Of course not, thanks to you. Include scripts and debug output, and maybe we'll get closer. Just tell me we're not doing your homework for you. > -Original Message- > From: Leo Burd [mailto:[EMAIL PROTECTED] > Sent: Monday, August 08, 2005 2:48 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] URGENT: Problems with PHP AGI... > > > Hello everyone, > > I'm having all sorts of problems with my PHP AGI scripts... > Basically, > my scripts run fine from the command line and don't do anything well > called from Asterisk. Here are my questions: > > a) Does Asterisk require PHP CLI or CGI? From the command line, my > script seems to work fine with PHP 4.3.11 (cli) but not with > PHP 4.3.9 (cgi) > > b) How to debug my script? According to the Asterisk CLI > messages, it > seems to be called, but nothing happens... > > c) Any ideas of what's going on? > > Thanks in advance for any help, > > Leo > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO definition
Maybe I am to sensetive, but what is an FXO? I have a device in my hand, it says it has an FXS and FXO port (besides WAN and LAN port) The SIP settings are only effecting the FXS. The FXO is connected to the phone company but can only be reached from the phone connected to FXS by prepending a defined key (e.g. #) The FXO port is directly connected to FXS if the box is without power. A call from the phone company line will be directly connected to the phone on FXS. Can I still say it is an FXO port bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit <[EMAIL PROTECTED]> wrote: > Hi everybody, > > I need a little clarification regarding the hardware to be used with > asterisk. I want to setup an asterisk box to make calls through both > internet and pstn, but i heard frm my friend (he was not sure) that digium > cards are incompatible with indian telephony systems, is it so? If yes, then > is there a way around this problem? > > Thanks in advance, > Ankit > > P.S- It would be greatly appreciated if someone could provide a technical > explanation to why digium cards are incompatible with indian (or anyother > telephone system), i thought telephone network is same everywhere. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue-hold time + weight in astersk+acd
Hello list, There seem to be some problem with the ACD of asterisk where when we use this parameter in queues.conf . We could not get any announcement as expected. Iam useing the latest CVS-head Even weight also doesnot seem to work properly I tried like this where we have two queues one with 100 weight and another with 200 as weight when both enter into the queue when queue is empty when agent logs is who is member in both teams the first call connected is the one which first enter into queue irrestpective of the weights so any ideas. my queues.conf file is as follows [queue1] member => Agent/1000;rk1 member => Agent/1001;rk2 announce-holdtime=yes announce-frequency=90 weight=100 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-minutes = queue-minutes queue-thankyou = queue-thankyo [queue2] member => Agent/1000;rk1 member => Agent/1001;rk2 announce-holdtime=yes announce-frequency=90 weight=200 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-minutes = queue-minutes queue-thankyou = queue-thankyou with regards rkvalmiki __ How much free photo storage do you get? Store your friends 'n family snaps for FREE with Yahoo! Photos http://in.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] function declaration isn't a prototype
hi dave, any suggestions on myencoutrered problem below? thnks so much. chris - Original Message - From: "chris" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, August 08, 2005 4:51 PM Subject: Re: [Asterisk-Users] function declaration isn't a prototype > hi dave, > > yes, it was indeed missing, so i downloaded ncurses from sunfreeware.com > and got a new error, > > /usr/local/sparc-sun-solaris2.8/bin/ld: cannot find -lssl > > so i downloaded openssl-0.9.7g-sol9-sparc-local.gz from sun freeware,it was > installed in /usr/local/ssl/.. the compile did contiue, however, i have a > new error. > > > > else \ > mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ > fi > > rm -f include/asterisk/version.h.tmp > make[1]: `ast_expr.a' is up to date. > make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk' > gcc -g -o asterisk io.o sched.o logger.o frame.o loader.o config.o > channel.o t ranslate.o file.o > say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmod > em.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o dsp.o > chanvars.o indications.o > autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt. > o aestab.o aeskey.o utils.o config_old.o plc.o jitterbuf.o dnsmgr.o > devicestate. o netsock.o > slinfactory.o strcompat.o ast_expr.a editline/libedit.a db1-ast/libd > b1.a > stdtime/libtime.a -lncurses -lm -lpthread -ldl -lnsl -lsocket -lresolv -L/u > sr/local/ssl/lib -lssl > /usr/local/sparc-sun-solaris2.8/bin/ld: warning: libcrypto.so.0.9.7, needed > by / > usr/local/ssl/lib/libssl.so, not found (try using -rpath or -rpath-link) > /usr/local/sparc-sun-solaris2.8/bin/ld: warning: libgcc_s.so.1, needed by > /usr/l ocal/ssl/lib/libssl.so, > not found (try using -rpath or -rpath-link) > utils.o: In function `vasprintf': > /export/home/fst/chris/cvs/asterisk/utils.c:623: undefined reference to > `va_copy ' > /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_find_type' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_enc_null' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_CIPHER_CTX_init' > /usr/local/ssl/lib/libssl.so: undefined reference to `X509_NAME_dup' > /usr/local/ssl/lib/libssl.so: undefined reference to `COMP_compress_block' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc2_cbc' > /usr/local/ssl/lib/libssl.so: undefined reference to `sk_new_null' > /usr/local/ssl/lib/libssl.so: undefined reference to > `X509_STORE_get_by_subject' > /usr/local/ssl/lib/libssl.so: undefined reference to `lh_free' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_VerifyFinal' > /usr/local/ssl/lib/libssl.so: undefined reference to `COMP_CTX_new' > /usr/local/ssl/lib/libssl.so: undefined reference to `sk_dup' > /usr/local/ssl/lib/libssl.so: undefined reference to > `X509_STORE_CTX_set_ex_data ' > /usr/local/ssl/lib/libssl.so: undefined reference to `X509_free' > /usr/local/ssl/lib/libssl.so: undefined reference to `CRYPTO_get_ex_data' > /usr/local/ssl/lib/libssl.so: undefined reference to `BN_bin2bn' > /usr/local/ssl/lib/libssl.so: undefined reference to > `CRYPTO_get_ex_new_index' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_DigestFinal_ex' > /usr/local/ssl/lib/libssl.so: undefined reference to > `PEM_read_bio_RSAPrivateKey ' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_aes_128_cbc' > /usr/local/ssl/lib/libssl.so: undefined reference to `BN_bn2bin' > /usr/local/ssl/lib/libssl.so: undefined reference to `RAND_add' > /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_s_socket' > /usr/local/ssl/lib/libssl.so: undefined reference to `asn1_add_error' > /usr/local/ssl/lib/libssl.so: undefined reference to `BUF_MEM_grow_clean' > /usr/local/ssl/lib/libssl.so: undefined reference to `d2i_RSAPrivateKey' > /usr/local/ssl/lib/libssl.so: undefined reference to `sk_num' > /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_free_all' > /usr/local/ssl/lib/libssl.so: undefined reference to `RSA_up_ref' > /usr/local/ssl/lib/libssl.so: undefined reference to `BIO_get_retry_reason' > /usr/local/ssl/lib/libssl.so: undefined reference to `X509_STORE_new' > /usr/local/ssl/lib/libssl.so: undefined reference to `HMAC_Final' > /usr/local/ssl/lib/libssl.so: undefined reference to `HMAC_Init_ex' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_md5' > /usr/local/ssl/lib/libssl.so: undefined reference to `ASN1_object_size' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_get_cipherbyname' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc4' > /usr/local/ssl/lib/libssl.so: undefined reference to `EVP_ad
[Asterisk-Users] Re: info regarding hardware
hi rajesh, thx for the info., also did u purchase the card directly frm digium or there are distributors of digium cards in india. -ankit On 8/9/05, rajeshkumar nayak <[EMAIL PROTECTED]> wrote: > hi > > Digium card is compatible with the indian telephone line.I am currently > using TDM400P(TDM11B) and this is working fine.It has one FXS and one FXO > interface . > > rajesh > > Ankit <[EMAIL PROTECTED]> wrote: > Hi everybody, > > I need a little clarification regarding the hardware to be used with > asterisk. I want to setup an asterisk box to make calls through both > internet and pstn, but i heard frm my friend (he was not sure) that digium > cards are incompatible with indian telephony systems, is it so? If yes, then > is there a way around this problem? > > Thanks in advance, > Ankit > > P.S- It would be greatly appreciated if someone could provide a technical > explanation to why digium cards are incompatible with indian (or anyother > telephone system), i thought telephone network is same everywhere. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > - > Start your day with Yahoo! - make it your home page > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay problem
Hi, I've experienced excessive delay when called from one extension number to another... This happened unstable, as the delay range between 2 - 20 seconds... I'm using Duron 950 MHz with memory 256 MB as asterisk server and my asterisk currently serves 30-40 accounts.. Concurrent calls vary between 1-10 calls. Is my Duron overwhelmed by the load? The delay exists in queue, local sip-to-sip call, and zap-to-sip call. It's so annoying :( Anyone has a solution or maybe some clue for me? Totally clueless here... Thanks... Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS - Don't want a Dailtone
Robert Christian wrote: Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? You have come across one of the few times you want immediate=yes. When the phone is picked up Asterisk will try exten => s in the context that channel is in. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Anyone having issues with sipphone?
All of a sudden, my account doesn't appear to work, or even perhaps exist with SIPPhone. Is anyone else having trouble? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO gateways / Audiocodes MP-108
My experience with an MP-108 was similar. Incredibly complex to setup, and very little help from MFR, or even ABPTECH, the main US reseller. We just couldn't get it working properly. Ended up with a TE110P with an Adit 600 channel bank, which ROCKS. Unbelieveably easy to setup. No echo whatsoever. -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
I'll give it a shot.. Do you know if they have any plans to merge this in? On 8/8/05, Gary Reuter <[EMAIL PROTECTED]> wrote: > On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote: > > I guess the problem is with SIPPhone then. I opened a ticket with > > them. I'll post their response when I have one. > > > I wouldn't bet money on that yet... > I've seen identical DTMF problems (doubled and mangled) digits and > I've never used SIPPhone. > I had traced it to a problem in the way asterisk handles > out-of-sequence RFC2833 dtmf indications. See the -dev thread here: > http://lists.digium.com/pipermail/asterisk-dev/2005-May/thread.html#12655 > > I tracked it down by using ethereal on the sip and rtp streams between > my different (working and non-working) SIP providers. The only > difference I could find was the order of the dtmf packets, and with > the simple change to an 'if' statement mentioned in the -dev thread, > the problems all went away. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P with Caller-ID in Australia,
I'll bet a slab -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Craig Sent: Tuesday, August 09, 2005 11:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] X100P with Caller-ID in Australia, You require no changes to detect caller id with a x100p card in Au, however most carriers only provide cid as an extra cost option. Your most likely problem is CID is not being sent to you. If you believe you have the caller id being sent to you, Find a caller id box/phone and confirm it is there. In 99% of cases where I have seen people in Au complaining of no caller id, it was because they didn't have the option enabled or because they requested it but the carrier hadn't actually enabled it. I'll bet a beer on it, you can't blame the card if the CID is not there for it to detect. cr Message: 27 Date: Tue, 9 Aug 2005 10:28:27 +0930 From: Justin Hawkins <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] X100P with Caller-ID in Australia, anyone? To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On 8/8/05, Jon Whitear <[EMAIL PROTECTED]> wrote: > > >Most likely your current card will work in Australia, but you need to > >patch the Asterisk Source to support the Australian Caller ID standard. > > > Yes, I've done that (and just to make sure, I've just upgraded to 1.0.9, > and applied that change.) Still no go, I'm afraid - I still get:- Hi Jon, I know this is not particularly helpful to you, but for what it's worth, I am in Australia, and get caller-id OK with an X100P clone and an unpatched Asterisk (FreeBSD ports). - Justin -- Justin Hawkins | [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 13, Issue 56 ** ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO gateways
> > Do you use this? Are you happy with it's performance? My experience a > > year ago with other small FXOs was very dissappointing. I tried the > > very earliest SPA-3000, TDM-400 and X101p. None were satisfactory so > > resorted to call forwarding my two POTS lines to a lines from an ITSP. > Nope, I don't. I use a TDM-400. But I've seen good comments about > Mediatrix's product on this list. Hard to configure but once it's > done, "It just Work" (tm). Maybe someone who actually use it can > comment on it. I'm not the OP, but did eval the 1204 about a year ago. Worked very well in terms of transmission levels, echo canceller, etc. But, it was not at all easy to configure and the only support that exists is through resellers (which as been almost non-existant). Unless their firmware has changed a lot (in the last year), security was non-existant as anyone with even limited knowledge could hack the config (or place calls through it). The 1204 does not have a sip 'register' function, and getting it to work with asterisk required some very unusual methods. However, once working, it did a very good job. On going firmware upgrades are not available to anyone without paying a fee, making the unit one of the most expensive to own. To bad the audio quality and echo canceller can't be moved to the TDM card. :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] URGENT: Problems with PHP AGI...
Leo Burd wrote: Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: Probably because it Asterisk is trying to give you info. a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According to the Asterisk CLI messages, it seems to be called, but nothing happens... c) Any ideas of what's going on? Are you reading back the data from Asterisk? Are you flushing the buffers? Have a search for this question on google. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote: > I guess the problem is with SIPPhone then. I opened a ticket with > them. I'll post their response when I have one. > I wouldn't bet money on that yet... I've seen identical DTMF problems (doubled and mangled) digits and I've never used SIPPhone. I had traced it to a problem in the way asterisk handles out-of-sequence RFC2833 dtmf indications. See the -dev thread here: http://lists.digium.com/pipermail/asterisk-dev/2005-May/thread.html#12655 I tracked it down by using ethereal on the sip and rtp streams between my different (working and non-working) SIP providers. The only difference I could find was the order of the dtmf packets, and with the simple change to an 'if' statement mentioned in the -dev thread, the problems all went away. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
Chris Mason wrote: JP Carballo wrote: Chris Mason (Lists) wrote: Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. Did you find any noise, hum or gating on the FXO ports on incoming calls? Nope. Crystal clear calls. That's PSTN -> VG-400 -> Phone and PSTN -> VG-400 -> * -> VG-400 -> Phone Any idea where I can ge them? I replied to your query earlier Re: FXO Gateways with a quote. Feel free to write to me for specifics. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM Hint for MeetMe
Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS - Don't want a Dailtone
Robert Christian wrote: Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert The famous batphone mode :) Aside from the aforementioned zapata and immediate mode, take a look at AgentLogin. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
I guess the problem is with SIPPhone then. I opened a ticket with them. I'll post their response when I have one. Thanks! -JD- On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for > my purposes. I've been really pounding on our sipphone number the past half > hour or so and I'm seeing the same issues you are. Sometimes it hits > correctly, sometimes it doesn't. IE, Dialing 5954, some of the times it > works, sometimes it's 5594, 5994, 5944, 59, 95, 54, etc. I know I'm not > fatfingering the dialing because my cell prints the dtmf digits to the > screen. We haven't been seeing the issues here because our sipphone number > isn't published (yet). > > Louie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
Steve Underwood wrote: JP Carballo wrote: I'll post the VG-400 settings here and on voip-info.org as soon as I get my notes together. Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. I haven't used the VG-400, but I have used their 1 port unit. The only hardware problem I had with that was because its construction is *not* sturdy. They don't strap down some fairly heavy inductors on the PCB. During transit, bumping had cause the inductors to break off. After soldering them back the machine worked OK. The unit's T.38 code is buggy. It seems it can work in some setups, but in others the machine locks up and needs a hard reset when it hears FAX tone. Regards, Steve This? http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG100TA In which case, I agree. It's plastic and the whole case is held together by a single screw in the center, while the VG-400's case is of metal. I still have like a dozen or so in stock. In fact, one of the VG-100's got shipped back to us with a hole in it. Go figure. Some rat I'd wager. I'll keep the T.38 factoid in mind when we need fax services. That does not bode well though. Yoda may well have used the same libraries for T.38 support in their product line. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with cmd monitor
Was using this monitor line to get soxmix to mix test-in.wav and test-
out.wav into test.wav.
exten => 1200,1,Monitor(wav|/tmp/test|m)
When I start the conference, the * console shows this:
monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test-
out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) &
/tmp shows test-in.wav, test-out.wav and test.wav. From what I can
see above, it appears as though the last command the m option runs
seems to be wrong which is why the in and out aren't being deleted.
I changed it to this:
exten => 1200,1,SetVar(MONITOR_EXEC=/bin/nice -n 19 /usr/bin/soxmix)
exten => 1200,2,SetVar(MONITOR_EXEC_ARGS=&& /bin/rm /tmp/test-*)
exten => 1200,3,Monitor(wav|/tmp/test|m)
exten => 1200,4,MeetMe(37455,APM)
This is what * console looks like:
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/3", "MONITOR_EXEC=/usr/bin/
soxmix") in new stack
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/3", "MONITOR_EXEC_ARGS=&& /bin/
rm /tmp/test-*") in new stack
-- Executing Monitor("IAX2/[EMAIL PROTECTED]/3", "wav|/tmp/test|m") in new
stack
-- Executing MeetMe("IAX2/[EMAIL PROTECTED]/3", "37455|APM") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '37455'
-- Playing 'conf-getpin' (language 'en')
-- Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/3
Aug 8 21:20:32 WARNING[19742]: app_meetme.c:962 conf_run: Unable to
write frame to channel: Resource temporarily unavailable
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/3
-- Hungup 'Zap/pseudo-1886765963'
== Spawn extension (drillsquad-conf, 1200, 4) exited non-zero on
'IAX2/[EMAIL PROTECTED]/3'
-- Hungup 'IAX2/[EMAIL PROTECTED]/3'
monitor executing /usr/bin/soxmix "//tmp/test-in.wav" "//tmp/test-
out.wav" "//tmp/test.wav" && /bin/rm /tmp/test-* &
ricky*CLI>
It looks like it ran soxmix fine, but it didn't remove the -in and -
out files.
Anyone had any luck with this?
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Re: [Asterisk-Users] FXS - Don't want a Dailtone
On Monday 08 August 2005 21:19, Robert Christian wrote: > Does anyone know of a way to make a standard analog phone plugged into an > FXS port do something other than get a dialtone when you pick it up? For > example, if the phone should automatically ring someone or play a greeting > when picked up without having to enter an extension? Read the asterisk handbook. Pay particular attention to the section on zapata and immediate mode. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P with Caller-ID in Australia,
You require no changes to detect caller id with a x100p card in Au, however most carriers only provide cid as an extra cost option. Your most likely problem is CID is not being sent to you. If you believe you have the caller id being sent to you, Find a caller id box/phone and confirm it is there. In 99% of cases where I have seen people in Au complaining of no caller id, it was because they didn't have the option enabled or because they requested it but the carrier hadn't actually enabled it. I'll bet a beer on it, you can't blame the card if the CID is not there for it to detect. cr Message: 27 Date: Tue, 9 Aug 2005 10:28:27 +0930 From: Justin Hawkins <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] X100P with Caller-ID in Australia, anyone? To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On 8/8/05, Jon Whitear <[EMAIL PROTECTED]> wrote: > > >Most likely your current card will work in Australia, but you need to > >patch the Asterisk Source to support the Australian Caller ID standard. > > > Yes, I've done that (and just to make sure, I've just upgraded to 1.0.9, > and applied that change.) Still no go, I'm afraid - I still get:- Hi Jon, I know this is not particularly helpful to you, but for what it's worth, I am in Australia, and get caller-id OK with an X100P clone and an unpatched Asterisk (FreeBSD ports). - Justin -- Justin Hawkins | [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 13, Issue 56 ** ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS - Don't want a Dailtone
Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
JP Carballo wrote: Chris Mason (Lists) wrote: Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. Did you find any noise, hum or gating on the FXO ports on incoming calls? Nope. Crystal clear calls. That's PSTN -> VG-400 -> Phone and PSTN -> VG-400 -> * -> VG-400 -> Phone Any idea where I can ge them? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P with Caller-ID in Australia, anyone?
On 8/8/05, Jon Whitear <[EMAIL PROTECTED]> wrote: > > >Most likely your current card will work in Australia, but you need to > >patch the Asterisk Source to support the Australian Caller ID standard. > > > Yes, I've done that (and just to make sure, I've just upgraded to 1.0.9, > and applied that change.) Still no go, I'm afraid - I still get:- Hi Jon, I know this is not particularly helpful to you, but for what it's worth, I am in Australia, and get caller-id OK with an X100P clone and an unpatched Asterisk (FreeBSD ports). - Justin -- Justin Hawkins | [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 4.0 firmware issue
Colin E. McDonald wrote: The new update seems to have cured my issue with calls intersecting and Zap lines not being hung up after the user terminates the session but now I am having sound issues with all of my phones. The sounds seems to be very low on all of them and there is a definite change from the same set when it was at 3.6j. The speaker also generates what appears to be static but you can discern a scratchy sounding echo. This is also occuring on all phones after the upgrade. I have genereated a support ticket to Snom but I wanted to see if anyone on the list has run into the same behavior. Thanks Colin I have about 15 snom 360 phones loaded with 4.0 - and mine seem to be working great. I did update the memory manager as well, not sure if it helped with the issues you mention because I loaded it right after 4.0: http://snom.com/download/share/snom360-3.31-r.bin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
JP Carballo wrote: I'll post the VG-400 settings here and on voip-info.org as soon as I get my notes together. Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. I haven't used the VG-400, but I have used their 1 port unit. The only hardware problem I had with that was because its construction is *not* sturdy. They don't strap down some fairly heavy inductors on the PCB. During transit, bumping had cause the inductors to break off. After soldering them back the machine worked OK. The unit's T.38 code is buggy. It seems it can work in some setups, but in others the machine locks up and needs a hard reset when it hears FAX tone. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and .NET
> Are there any Asterisk interfaces with .NET? There is a port of the Manager API implementation of Asterisk-Java available for .NET from Chad Kitching. You can download it from http://www3.mb.sympatico.ca/~chadk/ =Stefan signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF issues with SIPPhone?
I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for my purposes. I've been really pounding on our sipphone number the past half hour or so and I'm seeing the same issues you are. Sometimes it hits correctly, sometimes it doesn't. IE, Dialing 5954, some of the times it works, sometimes it's 5594, 5994, 5944, 59, 95, 54, etc. I know I'm not fatfingering the dialing because my cell prints the dtmf digits to the screen. We haven't been seeing the issues here because our sipphone number isn't published (yet). Louie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > RFC2833 is sent out of band. What's the output on your asterisk console? I don't see any output during this time on my asterisk console. Unless there's additional logging I'd need to enable? Thanks for the help! -JD- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and .NET
Hi, Are there any Asterisk interfaces with .NET? Thanks, Alvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Press # to continue / Findme
I have implemented a simple findme solution based on DID's.
In the findme context, after trying each respective number (at s,5 and
s,6), I would like a voice saying "The person was not available, press
pound to try the next number." Otherwise, it hangs up after 20 seconds
without dialing the next number.
Any ideas? Using background dosen;t work, because you hit # and it
hangs up.
[default]
exten => _8134712509,1,Goto(columbia,s|1)
exten => _8134712510,1,Goto(constitution,s|1)
[columbia]
exten => s,1,setvar(GSMNUM=xx)
exten => s,2,setvar(IRINUM=xx)
exten => s,3,setvar(F55NUM=xx)
exten => s,4,macro(findme,${GSMNUM},${IRINUM},${F55NUM})
[constitution]
exten => s,1,setvar(GSMNUM=xxx)
exten => s,2,setvar(IRINUM=xxx)
exten => s,3,setvar(F55NUM=xxx)
exten => s,4,macro(findme,${GSMNUM},${IRINUM},${F55NUM})
[macro-findme]
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,BackGround(pls-wait-connect-call)
exten => s,4,Dial(SIP/[EMAIL PROTECTED],20,m)
exten => s,5,Background(gsm)
exten => s,6,Background(silence/5)
exten => s,7,Dial(Zap/1/${ARG2},15,m)
exten => s,8,Background(iridium)
exten => s,9,Background(silence/5)
exten => s,10,Dial(${ARG3}/sip.broadvoice.com,10,m)
exten => s,11,Hangup
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[Asterisk-Users] Re: asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards)
Morning, I've installed asterisk on FC4 and had a few problems with zaptel stuff. Having installed it on SuSE I was able to check a few things that were different. When installed with the RPMS, the udev stuff gets put into the normal 50-udev file under /etc/udev. On SuSE, it worked when the necessary zaptel section was in a separate file name 55-zaptel.udev (the name is arbitrary, however the number was the important part). So i'm presuming this is a udev matter. When i first started working with asterisk it was on mandrake with a 2.6 kernel - again, with this distro i had troubles with the udev and zaptel not working. From this went to SuSE and it was no probs. I am still wrestling with asterisk on FC4. The current problem is CDR dumped into the MySQL database, and AMP not working correctly. The problem is not being able to connect to the manager "Unable to connect to manager (13)" within AMP when making changes. I'm thinking it's a little PHP problem at the moment, however because I work on this box with another colleague and we both hack the sht out of configs and stuff, i can't be certain. >Hi, > >On Sat, Aug 06, 2005 at 09:05:51AM -0400, Zachary Whitley wrote: > On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote: > > Kumara Jayaweera wrote: > > >Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any > > >success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during > > >installation (FC4). Please any comments? > > Yes, Without problems. Can u install RH9 on ur box? > > I'm assuming that Madhawa is suggesting that you install RH9. I've > installed Asterisk on FC4 with very few problems. Start with a > standard FC4 installation then install the following rpms from > atrpms.net: > > asterisk-addons asterisk-sounds zaptel zaptel-devices > One little problem. Maybe it's been fixed but last time I checked it > wasn't. In the /etc/init.d/zaptel the path to ztcfg is > incorrect. Find all references to ztcfg and change them to => > /usr/sbin/ztcfg > > You can copy the sample configs from > /usr/share/doc/asterisk-1.0.9/configs/ to get you going. Running > asterisk -c -vvv will let you know which ones you need. > > The rest is going to be specific to your hardware and setup. Good luck. >Thanks for the comments on the packages. >I'm looking for more feedback and improvements on the asterisk and >friends rpms at ATrpms. Red Hat Linux, Fedora Core and RHEL (and >clones) are supported. > >There are already some bug reports at bugzilla.atrpms.net on >enhancements and bugs in the packages, see > >http://bugzilla.atrpms.net/buglist.cgi?query_format=advanced&short_desc_type=allwordssubstr&short_desc=&long_desc_type=substring&long_desc=asterisk&bug_file_loc_type=allwordssubstr&bug_file_loc=&bug_status=NEW&bug_status=ASSIGNED&bug_status=REOPENED&emailassigned_to1=1&emailtype1=substring&email1=&emailassigned_to2=1&emailreporter2=1&emailcc2=1&emailtype2=substring&email2=&bugidtype=include&bug_id=&votes=&chfieldfrom=&chfieldto=Now&chfieldvalue=&cmdtype=doit&order=Reuse+same+sort+as+last+time&field0-0-0=noop&type0-0-0=noop&value0-0-0= > >Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting hangup - TDM400P / X100P
The setting for Australia looks correct so that means I'm not setting the country code correctly, despite what I thought. How can I determine if Asterisk has the correct country code? (I did try to check this again on the Wiki but it's down at the moment :( tony David Phelan wrote: Have a look at the indications.conf file Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson Sent: Tuesday, 9 August 2005 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Detecting hangup - TDM400P / X100P I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to Australia and input it into the Sipura. How do I input these tones into the TDM400P as being a hangup? TIA, tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 4.0 firmware issue
The new update seems to have cured my issue with calls intersecting and Zap lines not being hung up after the user terminates the session but now I am having sound issues with all of my phones. The sounds seems to be very low on all of them and there is a definite change from the same set when it was at 3.6j. The speaker also generates what appears to be static but you can discern a scratchy sounding echo. This is also occuring on all phones after the upgrade. I have genereated a support ticket to Snom but I wanted to see if anyone on the list has run into the same behavior. Thanks Colin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help interpreting channel stats?
Could someone please look at this information and help me decipher what it should actually mean to me? I've found a bit of information here and there but I'd like to know what I'm supposed to be reading into this information: pbx*CLI> iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/teliax-4 208.139.204.228 opto-pps4/00014 00037/00039 00040ms 0004ms 0063ms g729 1 active IAX channel(s) pbx*CLI> iax2 show netstats LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/teliax-4 1045 62 213 0301 880 40 0 0 00 0 1 active IAX channel(s) I assume that the lost packets and drop packets are the key numbers here and at least partially explain the audio quality issues we're seeing. Are the other numbers OK? Why are almost all the remote numbers 0? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low sound
jonny hashem wrote: my customers complain that when they make a call they hear the another side very well but the another side hears the first side well but in low sound.what is the ptoblem here and i have to change? Could you be more vague? Try giving us hardware, relevant config sections, type of CO lines, someting to work with. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: OPAL now supports IAX2
What are the advantages of using woomera IAX2 instead of native IAX2?Put woomera aside right now, This is something that brings a cross platform IAX2 stack that can for example be used in Gnomemeeting or anything else that uses OPAL, using a closed and open familiar API. This can be used on windows, linux and anything that OPAL and PWLIB can be used on without any changes. Its a step in the right direction in my opinion./b ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > RFC2833 is sent out of band. What's the output on your asterisk console? I don't see any output during this time on my asterisk console. Unless there's additional logging I'd need to enable? Thanks for the help! -JD- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID
As I recall, should channels start as channel=>2 and not channel=2? I have all mine config'ed channel => 2 and it works fine... Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Saturday, August 06, 2005 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID > Ok, so the scope of the configuration is from channel= to channel= > statement with the configuration for the channel coming before the > channel statement. > > As in... > > > these=are > configs=for > the=first > channel=1 > > these=are > configs=for > the=second > channel=2 In fact, all the settings that you don't change stay the same as with the last channel. Ex.: callerid=asreceived otherparam=1 channel=2 otherparam=2 channel=3 In channel 3, the callerid param is still "asreceived" because nothing changed it hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan mapping for multiple outbound providers to determine best rates
Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may want to go through my pots line which has decent rates and good quality with an ld provider. My question is, with the dialplan, if you assign a number like 0114861. to terminate one provider, but do an include afterwards and include lets say, 011486135., will asterisk automatically route to the 011486135 or does it need to be specified in before the first include? Also, does anyone have any examples of how they route calls to the proper provider in extensions.conf, or must some extensive dialplan arrangements be assigned? The idea is simple, rather than having a huge extensions.conf with a lot of different exchanges, can asterisk determine the best route on its own based on includes, or must I have different lines in extensions.conf for asterisk to determine it? The whole idea being, if broadvoice is down, I would want a backup, but not neccesarily specify 10 lines of scripting for each country code and area code. I would think this is an important thing for high volume usages to save money. Basically, do any of the modules allow this in a simple manner, for example, where you can use a database to specify rates per call, and have asterisk decide the best route. This is a similar idea to LCR (least cost routing) on normal pbx systems. Any advice would be nice, since I'm sure those users who use asterisk for more commercial purposes have figured our a way to do this... Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialplan mapping for multiple outbound providers to determine best rates
Good day, I would recommend using an LCR engine to do this. There is at least one listed in the wiki. I am also nearing completion of an lcr engine that integrates with ASTPP, asterisk billing software. It will be easy to setup once I get it working. :-) Darren Wiebe [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may want to go through my pots line which has decent rates and good quality with an ld provider. My question is, with the dialplan, if you assign a number like 0114861. to terminate one provider, but do an include afterwards and include lets say, 011486135., will asterisk automatically route to the 011486135 or does it need to be specified in before the first include? Also, does anyone have any examples of how they route calls to the proper provider in extensions.conf, or must some extensive dialplan arrangements be assigned? The idea is simple, rather than having a huge extensions.conf with a lot of different exchanges, can asterisk determine the best route on its own based on includes, or must I have different lines in extensions.conf for asterisk to determine it? The whole idea being, if broadvoice is down, I would want a backup, but not neccesarily specify 10 lines of scripting for each country code and area code. I would think this is an important thing for high volume usages to save money. Basically, do any of the modules allow this in a simple manner, for example, where you can use a database to specify rates per call, and have asterisk decide the best route. This is a similar idea to LCR (least cost routing) on normal pbx systems. Any advice would be nice, since I'm sure those users who use asterisk for more commercial purposes have figured our a way to do this... Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Screening Sip Calls - Record()
I've posted about this before, but it's been so long I thought I'd see
if there is a new solution (can't find anything on google or wiki)
I use the Record() app on my incoming zap calls to record a persons name
if their caller id is not in the db. After the name is recorded, the
call is parked and ParkAndAnnounce announces that a call is holding from
${SCREENNAME}. Works fine with Zap, but obviously, SIP blows right
through the Record(). Is there any way to accomplish this on the SIP
channels?
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Re: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID
> As I recall, should channels start as channel=>2 and not channel=2? > > I have all mine config'ed channel => 2 and it works fine... > > Greg Yes, thanks for the correction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI perl problem
We start asterisk under a detached screen with GNU screen. We use the 'L' flag to log to a text file if we want. Then we can "screen -r" to the actual asterisk CLI any time we want. /usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc http://www.gnu.org/software/screen/ MATT--- On 8/8/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: > Tzafrir Cohen wrote: > > On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote: > > > >>Anish Basu wrote: > >> > >>>Hi, > >>> > >>>For some reason, my AGI perl scripts cannot write to the CLI console using > >>>standard error. I ran the agi-test.agi test script that came with asterisk > >>>and verified that the problem was not with the code. Asterisk is always > >>>started with 4 or more v's, yet this the CLI output does not show up. Have > >>>there been any major changes to AGI in the cvs-head? I am using Asterisk > >>>CVS-D2005.08.05. > >> > >>This is a known problem. Asterisk will only send STDERR from AGI > >>scripts to the actual console Asterisk is running on. You have to > >>switch to the actual console Asterisk is running on (tty9 I think) or > >>start asterisk in the foreground with "asterisk -cvvv". > >> > >>It's a minor issue for most people. > > > > > > And if you run with no "default terminal"? > > > > Any simple way around this in logger.conf (you can always tail -f the > > log file) > > > > logger.conf worked for me for everything EXCEPT AGI scripts that print > to STDERR. > > -- > Eric Wieling * BTEL Consulting * 504-210-3699 x2120 > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX TO IAX call between two registered servers
On Monday August 08 2005 7:06 pm, Carlos Chavez wrote:
> On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
> > Hello all,
> > I know this has been covered on list but can not find the answer I need,
> > lots of references to no authority found, but none with an answer.
> > I have two * servers, one behind firewall with nat the other on a dmz
> > with nat. Both servers register with each other successfully.
> > home is today's CVS-HEAD
> > away is Asterisk 1.0.7
> > on away: Registered to '165.xxx.xxx.xxx', who sees us as
> > 69.xxx.xxx.xxx:4569 on home: Registered IAX2 to '69.xxx.xxx.xxx', who
> > sees us as
> > 165.xxx.xxx.xxx:4569
> > When i place a call from home to away:
> > exten => _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]);
>
> I guess what you are trying to do here is dial 998 and then the remote
> extension number? If so your extension shoud be something like:
>
> exten => _998.,1,Dial(IAX2/x:[EMAIL PROTECTED]/${EXTEN:3})
>
I was actually just trying to get it to fall into the default context which is
set up as follows
[default]
exten => s,1, agi,voicemail.cpp|${CALLERIDNUM}; does a db lookup
exten => s,2, GoToIf($[${MAILUSER} = 0]?5:5);
exten => s,3, GoToIf($[${MAILUSER} = 1]?4:5);
exten => s,4, HasNewVoicemail([EMAIL PROTECTED]:INBOX)
exten => s,5, Dial(sip/577,20); Ring the phone on the sipura
exten => s,6, GoTo(myvoicemail,9002,1);
exten => s,105, GoTo(myvoicemail,8002,1);
so that the phone on the sipura should ring since voicemail.cpp will not find
a listed calleridnum. am i totally missing something?
> > I get the following message on home:
> > Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call
> > rejected by 69.xxx.xxx.xxx: No authority found
> >
> > and get this message on away
> > Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected
> > connect attempt from 165.xxx.xxx.xxx
> >
> > home iax.conf
> > [away]
> > type=peer
> > username=away
> > auth=plaintext
> > secret=x
> > host=dynamic
> > context=pap2
> > dissallow=all
> > allow=ulaw
> >
> > [away-in]
> > type=peer
> > auth=plaintext
> > secret=x
> > host=dynamic
> > context=pap2
> > dissallow=all
> > allow=ulaw
> >
> > [away-out]
> > type=peer
> > secret=x
> > username=away
> > host=dynamic
> > disallow=all
> > allow=ulaw
> >
> >
> > away iax.conf
> > [home]
> > type=peer
> > user=home
> > secret=x
> > host=dynamic
> > context=default
> >
> > [home-in]
> > type=user
> > username=home
> > secret=x
> > context=default
> >
> > [home-out]
> > type=peer
> > secret=x
> > username=home
> > host=my.domain.com
> >
> > any suggestions would be greatly appreciated.
> > Thank you,
> > John M
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RE: [Asterisk-Users] DTMF issues with SIPPhone?
RFC2833 is sent out of band. What's the output on your asterisk console? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? So the way I understand this is with rfc2833, DTMF is sent out of band. So does this mean that SIPPhone is interpreting the tones incorrectly? Asterisk shouldn't be doing any actual tone detection with this method, right? On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > Yes we are. I just double checked our line, and oddly, the dtmf tones aren't > getting sent to our asterisk server. Switched it back to rfc2833, and it > works. It was the other way around when I first connected us. Some informal > testing just now doesn't show the DTMF tone problem in rfc2833 mode that > you're having. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: DTMF issues with Vonage forwarded lines
Since Thursday (Aug/4/2005) my Vonage line that is fowrwarded (not even using their ATA) to a number that rings to a PRI that is connected to asterisk stop passing on DTMF, anybody having this issue? When calling directly the asterisk box, or forwarding any other lines, there are no problems. Just the Vonage line has this problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
So the way I understand this is with rfc2833, DTMF is sent out of band. So does this mean that SIPPhone is interpreting the tones incorrectly? Asterisk shouldn't be doing any actual tone detection with this method, right? On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > Yes we are. I just double checked our line, and oddly, the dtmf tones aren't > getting sent to our asterisk server. Switched it back to rfc2833, and it > works. It was the other way around when I first connected us. Some informal > testing just now doesn't show the DTMF tone problem in rfc2833 mode that > you're having. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about agent queuing in Asterisk
Hi: In our existing call center, we defined agents in different tasks, some of them are assigned as primary for a given task, for other tasks as overflow, which we want agents to work with some projects flexibly. Does Asterisk queuing can handle this kind of routing mechanism? Thanks, Tielin Xu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX TO IAX call between two registered servers
On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
> Hello all,
> I know this has been covered on list but can not find the answer I need, lots
> of references to no authority found, but none with an answer.
> I have two * servers, one behind firewall with nat the other on a dmz with
> nat. Both servers register with each other successfully.
> home is today's CVS-HEAD
> away is Asterisk 1.0.7
> on away: Registered to '165.xxx.xxx.xxx', who sees us as 69.xxx.xxx.xxx:4569
> on home: Registered IAX2 to '69.xxx.xxx.xxx', who sees us as
> 165.xxx.xxx.xxx:4569
> When i place a call from home to away:
> exten => _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]);
I guess what you are trying to do here is dial 998 and then the remote
extension number? If so your extension shoud be something like:
exten => _998.,1,Dial(IAX2/x:[EMAIL PROTECTED]/${EXTEN:3})
>
> I get the following message on home:
> Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call rejected
> by
> 69.xxx.xxx.xxx: No authority found
>
> and get this message on away
> Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect
> attempt from 165.xxx.xxx.xxx
>
> home iax.conf
> [away]
> type=peer
> username=away
> auth=plaintext
> secret=x
> host=dynamic
> context=pap2
> dissallow=all
> allow=ulaw
>
> [away-in]
> type=peer
> auth=plaintext
> secret=x
> host=dynamic
> context=pap2
> dissallow=all
> allow=ulaw
>
> [away-out]
> type=peer
> secret=x
> username=away
> host=dynamic
> disallow=all
> allow=ulaw
>
>
> away iax.conf
> [home]
> type=peer
> user=home
> secret=x
> host=dynamic
> context=default
>
> [home-in]
> type=user
> username=home
> secret=x
> context=default
>
> [home-out]
> type=peer
> secret=x
> username=home
> host=my.domain.com
>
> any suggestions would be greatly appreciated.
> Thank you,
> John M
>
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--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001
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[Asterisk-Users] IAX TO IAX call between two registered servers
Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees us as 69.xxx.xxx.xxx:4569 on home: Registered IAX2 to '69.xxx.xxx.xxx', who sees us as 165.xxx.xxx.xxx:4569 When i place a call from home to away: exten => _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]); I get the following message on home: Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call rejected by 69.xxx.xxx.xxx: No authority found and get this message on away Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect attempt from 165.xxx.xxx.xxx home iax.conf [away] type=peer username=away auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [away-in] type=peer auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [away-out] type=peer secret=x username=away host=dynamic disallow=all allow=ulaw away iax.conf [home] type=peer user=home secret=x host=dynamic context=default [home-in] type=user username=home secret=x context=default [home-out] type=peer secret=x username=home host=my.domain.com any suggestions would be greatly appreciated. Thank you, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Detecting hangup - TDM400P / X100P
Have a look at the indications.conf file Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson Sent: Tuesday, 9 August 2005 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Detecting hangup - TDM400P / X100P I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to Australia and input it into the Sipura. How do I input these tones into the TDM400P as being a hangup? TIA, tony Zero Effort Networking Pty Ltd ABN 38 082 434 446 PO Box 6045 Blacktown NSW 2148 www.zeroeffortnetworking.com.au [EMAIL PROTECTED] Tel: (02) 9676 3541 Fax: (02) 8569 2012 Message from: [EMAIL PROTECTED] Message to: asterisk-users@lists.digium.com Attached files: 0 This message contains confidential information and is intended for asterisk-users@lists.digium.com If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 7/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
Chris Mason (Lists) wrote: Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. Did you find any noise, hum or gating on the FXO ports on incoming calls? Nope. Crystal clear calls. That's PSTN -> VG-400 -> Phone and PSTN -> VG-400 -> * -> VG-400 -> Phone -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting hangup - TDM400P / X100P
I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to Australia and input it into the Sipura. How do I input these tones into the TDM400P as being a hangup? TIA, tony Zero Effort Networking Pty Ltd ABN 38 082 434 446 PO Box 6045 Blacktown NSW 2148 www.zeroeffortnetworking.com.au [EMAIL PROTECTED] Tel: (02) 9676 3541 Fax: (02) 8569 2012 Message from: [EMAIL PROTECTED] Message to: asterisk-users@lists.digium.com Attached files: 0 This message contains confidential information and is intended for asterisk-users@lists.digium.com If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN D-Channel Problem / bristuff / qozap
Hi, i am using a HFC-4S Board with the bristuff patches from Junghans. 3 of 4 ISDN-NTs working well but the first one of this group making troubles. (all in TE Mode.) every 10 minutes all active lines getting disconnected from this NT :-( asterisk showing up this information: == Primary D-Channel on span 2 down Aug 8 23:36:43 WARNING[7616]: chan_zap.c:2129 pri_find_dchan: No D-channels available! Using Primary on channel anyway 6! == Primary D-Channel on span 2 up Here are the debugging output from the qozap driver: ISDN NT1 (the bad one): Aug 8 23:16:30 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:16:30 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x1 0x1 0x53 0xae ] 6 bytes Aug 8 23:16:40 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:16:40 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x1 0x1 0x53 0xae ] 6 bytes Aug 8 23:16:47 voip kernel: qozap: card 2 span 2 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff 0xd9 0x70 ] 10 bytes Aug 8 23:16:48 voip kernel: qozap: card 2 span 2 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff 0xd9 0x70 ] 10 bytes Aug 8 23:16:49 voip kernel: qozap: card 2 span 2 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x6 0x1 0x98 0x5d ] 10 bytes Aug 8 23:16:49 voip kernel: qozap: card 2 span 2 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x6 0x1 0x98 0x5d ] 10 bytes Aug 8 23:16:50 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:16:53 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:16:56 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:16:59 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x7f ] 3 bytes Aug 8 23:17:02 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x7f ] 3 bytes Aug 8 23:17:02 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x73 0x8 0x9e ] 5 bytes Aug 8 23:17:12 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:17:12 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x1 0x1 0x53 0xae ] 6 bytes Aug 8 23:17:22 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:17:22 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x1 0x1 0x53 0xae ] 6 bytes Aug 8 23:17:32 voip kernel: qozap: card 2 stport 2 TX [ 0x0 0x1 0x1 0x1 ] 4 bytes Aug 8 23:17:32 voip kernel: qozap: card 2 span 2 RX [ 0x0 0x1 0x1 0x1 0x53 0xae ] 6 bytes ISDN NT2 (a good one): Aug 8 23:18:53 voip kernel: qozap: card 2 stport 1 TX [ 0x0 0x1 0x1 0xb ] 4 bytes Aug 8 23:18:53 voip kernel: qozap: card 2 span 1 RX [ 0x0 0x1 0x1 0x9 0x1b 0x22 ] 6 bytes Aug 8 23:19:03 voip kernel: qozap: card 2 stport 1 TX [ 0x0 0x1 0x1 0xb ] 4 bytes Aug 8 23:19:03 voip kernel: qozap: card 2 span 1 RX [ 0x0 0x1 0x1 0x9 0x1b 0x22 ] 6 bytes Aug 8 23:19:07 voip kernel: qozap: card 2 span 1 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff 0xd9 0x70 ] 10 bytes Aug 8 23:19:08 voip kernel: qozap: card 2 span 1 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff 0xd9 0x70 ] 10 bytes Aug 8 23:19:13 voip kernel: qozap: card 2 stport 1 TX [ 0x0 0x1 0x1 0xb ] 4 bytes Aug 8 23:19:13 voip kernel: qozap: card 2 span 1 RX [ 0x0 0x1 0x1 0x9 0x1b 0x22 ] 6 bytes Aug 8 23:19:23 voip kernel: qozap: card 2 stport 1 TX [ 0x0 0x1 0x1 0xb ] 4 bytes Aug 8 23:19:23 voip kernel: qozap: card 2 span 1 RX [ 0x0 0x1 0x1 0x9 0x1b 0x22 ] 6 bytes Aug 8 23:19:33 voip kernel: qozap: card 2 stport 1 TX [ 0x0 0x1 0x1 0xb ] 4 bytes Aug 8 23:19:33 voip kernel: qozap: card 2 span 1 RX [ 0x0 0x1 0x1 0x9 0x1b 0x22 ] 6 bytes interresing seems to me this line: Aug 8 23:16:49 voip kernel: qozap: card 2 span 2 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x6 0x1 0x98 0x5d ] 10 bytes does some know this problem, or can me point out how i can solve the problem ? where i can get more information what this bytes want to say me ? more asterisk debug information: < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 000 P/F: 1 < 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (0) > [ 00 01 01 01 ] > Supervisory frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000EA: 1 > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > N(R): 000 P/F: 1 > 0 bytes of data -- Restarting T203 counter < [ 00 01 01 01 ] < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 000 P/F: 1 < 0 bytes of data -- ACKing all packets from 0 to (Aug 8 23:46:25 WARNING[7616]: chan_zap.c:2129 pri_find_dchan: No D-channels available! Using Primary on channel anyway 6! T203 counter expired in weird state 2 Sending Set Asynchronous Balanced Mode Extended > [ 00 01 7f ] > Unnumbered frame
Re: [Asterisk-Users] Call Recording with *
On Mon, 2005-08-08 at 16:33 -0500, Eric Wieling aka ManxPower wrote: > Bryce Chidester wrote: > > On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote: > > > >>I'm attempting to set up call recording with Asterisk. Using > >> > >>automon => *1 ; One Touch Record > >> > >>in features.conf does not appear to be working. I'm using Polycom 501's > >>but when someone dials *1 while in a call, nothing happens. > >> > >>I'm wondering if the phone or Asterisk is even detecting the DTMF. I > >>suspect that is the problem but don't know how to verify or correct. > > > > > > > > Using Zaptel channels, I know it detects the DTMF (debug output says so) > > but nothing comes of it, or *0, *2, or any of the other feature codes. > > Call parking and # transfer work though, so I'm guessing they're simply > > not implemented yet, as of 1.0.8. > > > > They will never be put into 1.0.x since 1.0.x does NOT get new features. > It's bug fix only. > Wasn't expecting it to make it in if it wasn't already - merely identifying what version I was using. This isn't a big feature for me, and you can simply write a manger interface to turn on and off monitoring anyways. Just confirmation that this isn't in the 1.0.x branch -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc syslog flooding
Hi, my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this "normal"?: -- zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353, wanted 8 got 7), probably a buffer overrun. zaphfc: dropped audio (z1=2712, z2=2695, wanted 8 got 17, dropped 9). With Asterisk running (asterisk -vgc) but without ANY activity it prints these messages. Can anybody explain to me what the problem is? Or is there no "problem"?: -- -- parse_srv: SRV mapped to host sipgate.de, port 5060 Scheduled a registration timeout # 227 Stopping retransmission on '[EMAIL PROTECTED]' of Request 137: Found Registration successful Cancelling timeout 227 Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found -- parse_srv: SRV mapped to host sipgate.de, port 5060 Scheduled a registration timeout # 236 Stopping retransmission on '[EMAIL PROTECTED]' of Request 138: Found Stopping retransmission on '[EMAIL PROTECTED]' of Request 139: Found Registration successful Cancelling timeout 236 Thanks, Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FCC to require wiretaps from VoIP providers
Really nothing new. They've done this with wired carriers for years. -- Tom On 8/8/05, Adam Megacz <[EMAIL PROTECTED]> wrote: > > Scary. > > http://www.eff.org/news/archives/2005_08.php#003876 > > -- > PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380 > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.voip-info.org/ front page taken outby spammer
History page is back. I think it got too big for the software to deal with. I changed it to show only the last 100 versions. There is an 'undo' option on the history page, but its never worked correctly, and so I have not enabled it. I'm working on a software upgrade that will hopefully address some of these issues. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Remco Barende To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, August 08, 2005 10:36 AM Subject: Re: [Asterisk-Users] http://www.voip-info.org/ front page taken outby spammer >> >>> Today the front page of http://www.voip-info.org/ was taken out by a >>> spammer. It also seem the history page for http://www.voip-info.org/ was >>> also nuked. I've restored the best I could using google cache, but still >>> missing some information.>>> >>> Who is an admin on http://www.voip-info.org/ and can fix it?>> >> Google cache is a hard way to fix wiki-busting -- the easiest way is to click > on "history" at the top of the page, go right to the version before the spam, > copy it, then paste it into an edit of the page..>> Of course, now, it's harder, because since the page was restored, people have > since modified it.. (also, for some reason, when I click on "history", > nothing seems to happen)..Highly offtopic but weird that the wiki software doesn't have an option to undo all the changes that one user or one ip address made.Wouldn't be too hard to implement IMHO, just keep a copy of all the stuff that was changed / deleted / added for an x number of days and build an option to automagically undo the changes.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FCC to require wiretaps from VoIP providers
Scary. http://www.eff.org/news/archives/2005_08.php#003876 -- PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO gateways
Chris Mason (Lists) wrote: Has anyone found a suitable but not exorbitant 4-6 port FXO => sip gateway? I need something more compact than a channel bank and using many Sipura 3000s is a bit cumbersome. Suggestions? Yoda VG-400 325 USD or 183 QUID -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording with *
Bryce Chidester wrote: On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote: I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or correct. Using Zaptel channels, I know it detects the DTMF (debug output says so) but nothing comes of it, or *0, *2, or any of the other feature codes. Call parking and # transfer work though, so I'm guessing they're simply not implemented yet, as of 1.0.8. They will never be put into 1.0.x since 1.0.x does NOT get new features. It's bug fix only. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording with *
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote: > I'm attempting to set up call recording with Asterisk. Using > > automon => *1 ; One Touch Record > > in features.conf does not appear to be working. I'm using Polycom 501's > but when someone dials *1 while in a call, nothing happens. > > I'm wondering if the phone or Asterisk is even detecting the DTMF. I > suspect that is the problem but don't know how to verify or correct. Using Zaptel channels, I know it detects the DTMF (debug output says so) but nothing comes of it, or *0, *2, or any of the other feature codes. Call parking and # transfer work though, so I'm guessing they're simply not implemented yet, as of 1.0.8. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX and Realtime...
Is anyone using services like Voicepulse, Nufone or Sixtel with IAX Realtime? I simply cannot get those services to work no matter what I do. I already have voicemail and sip running from realtime, I also have a second Asterisk server that connects using IAX2 and that one works perfectly with Realtime. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. Did you find any noise, hum or gating on the FXO ports on incoming calls? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] howto let the stream not passing asterisk
Rosario Pingaro wrote: We need to configure asterisk to authenticate two sip ATAs, but the stream must go directly from one to another ata without tuching asterisk. Is this possible adding canreinvite=yes into sip.conf? is it true laso if asterisk doesn't recognize the spd (t38)? thanks Rosario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Yes, If they support the same codec and don't put "t" / "T" with Dial command on d extensions.conf. ATA186 has a problem with "canreinvite=yes" for more info http://lists.digium.com/pipermail/asterisk-doc/2004-June/000547.html Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] howto let the media stream not passing saterisk?
is there some one tha has bee able to passthrough t38 into asterisk? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI perl problem
Tzafrir Cohen wrote: On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote: Anish Basu wrote: Hi, For some reason, my AGI perl scripts cannot write to the CLI console using standard error. I ran the agi-test.agi test script that came with asterisk and verified that the problem was not with the code. Asterisk is always started with 4 or more v's, yet this the CLI output does not show up. Have there been any major changes to AGI in the cvs-head? I am using Asterisk CVS-D2005.08.05. This is a known problem. Asterisk will only send STDERR from AGI scripts to the actual console Asterisk is running on. You have to switch to the actual console Asterisk is running on (tty9 I think) or start asterisk in the foreground with "asterisk -cvvv". It's a minor issue for most people. And if you run with no "default terminal"? Any simple way around this in logger.conf (you can always tail -f the log file) logger.conf worked for me for everything EXCEPT AGI scripts that print to STDERR. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPPeersAction class file not foundintheAsterisk-java.jar file
You dont run applications using the Manager API as AGI scripts but as standalone Java applications. So in you case proably via java -cp asterisk-java-0.1.jar:. ManagerAPI =Stefan On Mon, 2005-08-08 at 16:31 +0530, Bharat M. Sarvan wrote: > Ok Mr. Stefan, > The contents of the file fastagi-mapping.properties are as > follows > > Hello.agi = ManagerAPI > > Where ManagerAPI is my class filename ManagerAPI.class. > > And the directory structure is as given below: > > /home/Bharat/AGISERVER/BharatJava/ > > Where the BharatJava directory holds the files > > fastagi-mapping.properties > asterisk-java-0.1.jar > and the .java files > > > > And the Asterisk server is running on Fedora Core 1. I start the asterisk > server using the command "asterisk -" at the command > prompt. > > So I was having doubt about the execution of the part of the code you have > sent me where we login into the Asterisk server using the class > ManagerConnection. > > Is there any way out so that I can issue the command other than using the > sendAction Method of the ManagerConnection to which I pass the object of the > class CommandAction. > > My asterisk server is already up and running. I just need to issue the > command using the CommandAction. But even if I run the sample code > "ManagerAPI" given on the link > http://asterisk-java.sourceforge.net/tutorial.html. I am getting the error > for "No script configured for agi://" > > Please do reply as to how I get through this problem... > > > > > > Regards, > Bharat M. Sarvan > Software Engineer - VoIP > EZZI BPO Pvt Ltd., > PUNE. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter > Sent: Saturday, August 06, 2005 7:02 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] SIPPeersAction class file not > foundintheAsterisk-java.jar file > > Hi, > > > I have all the necessary files for the code to be executed. The > > fastagi-mapping.properties file is also correct. But still I am getting > the > > error for > > > > The IP address is correct and as well as the agi file name. Does it make a > > difference giving a Tab or a space when giving the mapping of agi file > name > > and class file name in the fastagi-mapping.properties file. > > no that makes no difference > > > Is there any other reason for getting this error > > no > please post the contents of fastagi-mapping.properties, your directory > structure and the command you use to run java. > and please note: THERE IS NO WAY TO GET SIP PEERS VIA FASTAGI ANYWAY! > > =Stefan > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Recording with *
I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or correct. Any thoughts? Thanks Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI perl problem
On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote: > Anish Basu wrote: > >Hi, > > > >For some reason, my AGI perl scripts cannot write to the CLI console using > >standard error. I ran the agi-test.agi test script that came with asterisk > >and verified that the problem was not with the code. Asterisk is always > >started with 4 or more v's, yet this the CLI output does not show up. Have > >there been any major changes to AGI in the cvs-head? I am using Asterisk > >CVS-D2005.08.05. > > This is a known problem. Asterisk will only send STDERR from AGI > scripts to the actual console Asterisk is running on. You have to > switch to the actual console Asterisk is running on (tty9 I think) or > start asterisk in the foreground with "asterisk -cvvv". > > It's a minor issue for most people. And if you run with no "default terminal"? Any simple way around this in logger.conf (you can always tail -f the log file) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI perl problem
Anish Basu wrote: Hi, For some reason, my AGI perl scripts cannot write to the CLI console using standard error. I ran the agi-test.agi test script that came with asterisk and verified that the problem was not with the code. Asterisk is always started with 4 or more v's, yet this the CLI output does not show up. Have there been any major changes to AGI in the cvs-head? I am using Asterisk CVS-D2005.08.05. This is a known problem. Asterisk will only send STDERR from AGI scripts to the actual console Asterisk is running on. You have to switch to the actual console Asterisk is running on (tty9 I think) or start asterisk in the foreground with "asterisk -cvvv". It's a minor issue for most people. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where is the asterisk DB file stored?
I am using Asterisk CVS-D2005.06.24.04 and I am trying to figure out where the DB information such as DND and CFIM are stored. I checked /var/lib/asterisk/astdb, but the file seems to be the same size even after adding information to the DB. I would like to able to share this database across multiple servers. Currently, I have asterisk set up with res_mysql and the mysql database is replicated across a few servers. Being able to do the same for CFIM and DND info would be great. Any help or suggestions are greatly appreciated. Thanks, Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] howto let the stream not passing asterisk
We need to configure asterisk to authenticate two sip ATAs, but the stream must go directly from one to another ata without tuching asterisk. Is this possible adding canreinvite=yes into sip.conf? is it true laso if asterisk doesn't recognize the spd (t38)? thanks Rosario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
I'm sorry, I didn't take a close look at your conf files, so I dont know if this is your problem, (and anyway, I dont know enough about [EMAIL PROTECTED] to know if the same things is a problem there) however, it might be useful for others, sipgates documentation is "weak" over the issue of where to put your "register" line. It MUST go in the general section of your sip.conf, NOT (as seems to be indicated by their - and other peoples docs) next to the [sipgate] (or whatever) section... (unless, it so happens, that this is the first section after general of course!) The voip-info wiki is clear on the point (see sip.conf), Hope this helps :-) Cheers Mark. On 7 Aug 2005, at 22:46, Brian McCarey wrote: I've re-uploaded the config files in NON pdf Any help welcomed. Regards - Original Message - From: Brian McCarey To: asterisk-users@lists.digium.com Sent: Sunday, August 07, 2005 5:55 PM Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate. Hi all, I'm new to the forum. Oh nonewbie question coming, I hear you all cry! I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions beginning with *
Arik Funke wrote: can anybody tell me how to create an extension that starts with a *? The expression matching works well if * is embedded in numbers but if the extension starts with *, it is not executed but extension s instead. Is there another way besides using a lot of if statements in the s extension? This works for me on asterisk-1.0.9 dialong on an analog phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI perl problem
Hi, For some reason, my AGI perl scripts cannot write to the CLI console using standard error. I ran the agi-test.agi test script that came with asterisk and verified that the problem was not with the code. Asterisk is always started with 4 or more v's, yet this the CLI output does not show up. Have there been any major changes to AGI in the cvs-head? I am using Asterisk CVS-D2005.08.05. Thanks, Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.voip-info.org/ front page taken out by spammer
Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? Google cache is a hard way to fix wiki-busting -- the easiest way is to click on "history" at the top of the page, go right to the version before the spam, copy it, then paste it into an edit of the page.. Of course, now, it's harder, because since the page was restored, people have since modified it.. (also, for some reason, when I click on "history", nothing seems to happen).. Highly offtopic but weird that the wiki software doesn't have an option to undo all the changes that one user or one ip address made. Wouldn't be too hard to implement IMHO, just keep a copy of all the stuff that was changed / deleted / added for an x number of days and build an option to automagically undo the changes. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Web Access Security
Hello, I am interested in offering the voicemail web access to faculty & staff as a part of my new Asterisk deployment, and was just wondiering about the security implications. Have there been exploits for it? Is it ready for primetime use? What kind of things should I be wary of about this? I appreciate all of your advice! Thanks - Don ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
With the lack of info on Yoda Communications in Taiwan and their hardware, I thought I'd post my experience. I got my hands on a few H.323 VG-400's and VG-100TA's. http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400 2 of the VG-400's were 2FXO/2FXS models. A couple were deployed to SE Asia, where we planned to offer our services. Originally, I ran a GnuGK server as per Yoda's instructions. It turned out that they were using GnuGK internally. They run their own server for clients to test and also have an embedded version of GnuGK for sale which they call GK Pro. Pretty soon, we got everything configured and running and had a couple of months of live testing. From the test runs, we found our clients needed a couple of additional services and GnuGK simply did not have the flexibility to provide them. We decided that for the final system, we would ask outside help. However, with one exception, each time we inquired from a company, we were encouraged to shift to SIP. What to do with the H.323 units? I asked Yoda and they provided us with the SIP firmware with which to upgrade the units! Setting up TFTP and upgrading the firmware was trivial and painless. Since then, I've installed * and though we've hit a couple of configuration issues, I haven't seen a show stopper yet. I didn't even have to deal with echoes like when we first configured the units using H.323. Everything's chugging along nicely and we've started running live tests to and from SE Asia again. After this phase, I'll delve into the H.323 support in *. We still have quite a few VG-100TA's which could not be programmed to use SIP. Besides, H.323 is still being used where we deploy. Through all this, just one thing surprised me. Yoda seemed to have never played with * inhouse like they did with GnuGK. They had tested with FWD and Bell (no idea what the actual site is), but not *. I'll post the VG-400 settings here and on voip-info.org as soon as I get my notes together. Overall, I'm happy. It has sturdy construction, standard features, and most of all works just fine with *. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple MWI on a single phone?
Bill Wesson wrote: Chris, I'm testing a Grandstream GXP-2000. It supports multiple MWI. Very nice! I didn't know about thatis there anything specific you have to do to associate a softbutton with a particular extension's voicemail so the MWI works? I didn't see anything about this in the wiki but I am looking over the user manual. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO gateways
Do you use this? Are you happy with it's performance? I am using the SPA-3000 with good results, I just want a more professional solution. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] URGENT: Problems with PHP AGI...
Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According to the Asterisk CLI messages, it seems to be called, but nothing happens... c) Any ideas of what's going on? Thanks in advance for any help, Leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.voip-info.org/ front page taken out by spammer
Johan Nordström wrote: Try send an email to [EMAIL PROTECTED] I did regarding an other issue and recieved an answer from [EMAIL PROTECTED] . Johan There are two major products that come out of Berkeley: LSD and UNIX. We don't believe this to be a coincidence. -- Jeremy S. Anderson Paul Belanger skrev: Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? Google cache is a hard way to fix wiki-busting -- the easiest way is to click on "history" at the top of the page, go right to the version before the spam, copy it, then paste it into an edit of the page.. Of course, now, it's harder, because since the page was restored, people have since modified it.. (also, for some reason, when I click on "history", nothing seems to happen).. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and PostgreSQL
Bastian Schern wrote: Hello everybody, now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to use PostreSQL instead of MySQL? Regards Bastian If you want to go thru the hassle of installing ODBC and all related stuff to run PSQL, sure you can. -Matthew P.S. stick with mysql. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to config voicemail with mysql?
res_mysql.conf controls the RealTime interface driver to MySQL. cdr_mysql.conf controls the MySQL CDR Addon. Are you running CVS-HEAD? Have you installed res_config_mysql.so? What happens when you type "realtime mysql status" ? Did you look in the debug log for errors? -Matthew Wei Kun wrote: Hi; I followed the online http://www.onlamp.com/lpt/a/3956 to configure voicemail. Now it works well with voicemail.conf and store voicemail as file. Now I want to try to test out storing voicemail within mysql database, but nothing is inserted into the table. It seems Asterisk still is following the viocemail.conf. Do I miss some config? I follow http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail In exconfig.conf voicemail => mysql,asterisk,voicemail_users And the table is in the database. The CDR can insert fine into this database, but nothing regarding voicemail. btw, what the relationship between res_mysql.conf and cdr_mysql.conf, looks like a lot of duplicated info. Thanks Kun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failed IAX Connection
Hi:
I unseccessfuly tried to place a call from one IAX
behind a nat through another IAX with a real IP. I got
the following error message on Asterisk Console:
Executing Dial("OSS/dsp", "IAX2/wassim/01)") in new
stack
-- Called wassim/01)
Call rejected by 195.112.214.98: No such
context/extensionad:
-- IAX2/wassim-1 is circuit-busy
Aug 8 10:13:38 NOTICE[21759]: chan_iax2.c:1556
iax2_destroy: Avoiding IAX destroy deadlock
Th context in IAX.conf behind the NAT is:
[wassim]
type=peer
host=xxx.xxx.xxx.xxx
context=default
username=chawki
secret=secret
And the extension context is:
exten => 01,1,Dial,IAX2/wassim/${EXTEN})
The IAX server with the real IP has the following
context in the IAX.conf file
[riad]
type=user
username=chawki
secret=secret
context=tele
And in the extensions.conf file, it has
[tele]
exten => 01,1,Answer
exten => 01,n,DeadAGI(astcc.agi)
exten => 01,n,Hangup
I don't know why the call is rejected and the
interpretation for the error:
No such context/extensionad:
Regards;
Chawki
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Re: [Asterisk-Users] info regarding hardware
hi Digium card is compatible with the indian telephone line.I am currently using TDM400P(TDM11B) and this is working fine.It has one FXS and one FXO interface . rajeshAnkit <[EMAIL PROTECTED]> wrote: Hi everybody,I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance,AnkitP.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql sock location
Wei Kun wrote: Hi; In case your * and mysql are running on the same machine, and you get error "Failed to connect to mysql database server ..." when using Asterisk with Mysql database, check the location of mysql.sock not /tmp/mysql.sock, but /var/lib/mysql/mysql.sock Regards Kun The location of your mysql.sock is completly configurable when you first install mysql. ./configure --with-sock=/some/path/to/mysql.sock If you run with the defaults, yea it won't goto /tmp/mysql.sock Most admins don't use default. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF issues with SIPPhone?
Yes we are. I just double checked our line, and oddly, the dtmf tones aren't getting sent to our asterisk server. Switched it back to rfc2833, and it works. It was the other way around when I first connected us. Some informal testing just now doesn't show the DTMF tone problem in rfc2833 mode that you're having. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone? Louie, On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband > and make sure you're using a ulaw connection. If you use a lossy codec, it > will scramble the DTMF tones. Are you using SIPPhone? When I use dtmfmode=inband, it just doesn't recognize the tones at all.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:
On Monday 08 August 2005 10:56, Kib Eki wrote:
Misunderstanding: I need to change the calleridnum because there is missing
the 0 before the number.
SetCIDNum(0${CALLERIDNUM}) or something?
Not to be nitpicky, but * will complain that SetCIDNum is deprecated.
SetVar(ZEROPREFIX=0)
Set(CALLERID(number)=${ZEROPREFIX}${CALLERIDNUM})
* will complain that SetVar is deprecated as well. But since I as yet
haven't found how to call the new SetVar, I just pretend I don't see
them. Anyone know?
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level,
I'm really quite busy.
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Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Kib Eki wrote: > >>2. A call made from a SIP client to the outside lacks the extension in the > >>number: Ex: PSTN number is 6789-0. The extension 234 is not added to the > >>PSTN number like 6789-234 when dialing out over the PSTN. > > > > > > Again, trivial dialplan stuff. Your sip.conf will have the callerid for > > each > > SIP client and you can append that information to the outgoing CID. > > > That is set correctly and works between sip clients. it is only a problem > when i > try to dial out over zap/g1. Most likely you and your provider are not in agreement on how the calling party number should be encoded (number of digits and which Type Of Number / Numbering Plan). The TON/NPI is set with the prilocaldialplan option. Make sure you send the expected number of digits. You may have to do a SetCallerId() before the dial. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Eric Wieling aka ManxPower wrote: > Peter Svensson wrote: > > See internationalprefix, nationalprefix etc in the file zapata.conf. > > Those options are only available in BRIStuff. They have been in HEAD for quite some time. The 1.0.x-releaes are note really usable in a lot of situations. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP600 Presence question
I am working with some Polycom IP600's and for outgoing calls from the phones, the presence features work fine. I am utilizing the 100,hint,SIP/100 for these calls. The problem that I am seeing is that any inbound calls that originate from a IVR in which the extension is dialed, does not show the exension busy on the other phone. Has anyone seen this before? Does anyone know of a workaround? Thanks, Scott BTW: I am aware of the presence bug with the IP Phones, and this is while the phones are actually monitoring. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Quality Issues
I am having quality problems on SIP bound calls made over the Zap channels. All Sip only calls (Cisco phone through Asterisk to another Sip device sound fine). Our setup looks like this: User --> Executone PBX --> Asterisk Server --> Router --> Internet The user is using a legacy handset that works with the Executone PBX and accesses the server using a button that calls up the trunk group connecting the legacy PBX to the Asterisk server. According to our Executone technician, the T1 card in the legacy PBX does *not* provide timing so I have the Digium T1 card as the primary sync source as follows: - /etc/zaptel.conf - span=1,1,0,d4,ami e&m=1-24 Our /etc/asterisk/zapata.conf file looks like this: - /etc/asterisk/zapata.conf - [trunkgroups] [channels] musiconhold=default busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes callgroup=1 pickupgroup=1 rxgain=-5 txgain=3 immediate=yes signalling=em_w context=zap-incoming group = 1 channel => 1-24 I get static/popping/clicking on all calls made over the incoming Zap channels (that is, the calls originating from the legacy handsets that trunk into the Asterisk server.) I know very little about the Executone PBX we integrate into, it is under support and these types of questions go unanswered with them as it is out of the scope of the support contract. On the Executone console I can see that I am getting several Blue Alarms. There doesn't seem to be any issues on the Asterisk side, no Red/Yellow alarms. Are there any timing settings I can tweak that will improve the call quality? Anyone familiar with the Executone hardware? Any help at all would be appreciated. Thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
Louie, On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband > and make sure you're using a ulaw connection. If you use a lossy codec, it > will scramble the DTMF tones. Are you using SIPPhone? When I use dtmfmode=inband, it just doesn't recognize the tones at all.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
